[OpenSIPS-Users] nat problems with one-way audio and forwarding

[prescott at wcoil.com]

I need some advice:
I have a test case that looks like this:
outside customer calls a phone number, number is busy.
opensips looks up the customer preference and forwards the busy call to another 
phone.
the first (busy) number is behind  a nat.
the second is not.
I am using rtpproxy for my media relaying on the nat side.
The problem is that when the 200 OK response ggets sent from the second phone 
picking up the call, opensips does not fix the sdp in the message.
This results in one-way audio on the call.
I am using opensips-1.5.3
I am attaching the sip trace for reference.
The way I do call forwarding is: look for the 486 busy response and then 
append_branch to the forwarded destination.
as you can see in the invite, the sdp information is retained, but the system 
doesn't seem to recognise the ok response as part of that sip transaction.
Any help or suggestions of where to look would be appreciated.
Thanks.

-- Kelly Prescott
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